I am using licode docker latest image on my machine to run and test licode video recording feature.
The recording feature is working as expected only audio is missing in recored *.mkv file.
I have already cross checked the configuration part. Audio in config is true while making the connection with erizo.
localStream.hasAudio() return the true. So i have no idea what i am missing here.
Related
I need to replay guacamole replay file by html5.
But the only thing I can do now is to use guacenc to convert the playback file into a video and play it locally.
I tried to build a demo based on the secondary development of guacamole-example to realize the playback of the screen file.
Although I still don't know the principle of its implementation, it solves my problem very well.
I also tried the official demo, but there is a problem with maven dependencies.
I'm working with libffmpeg in an iOS app. My goal is to connect to an RTSP source and write the media out to a file that can later be used with the iOS media player. Ideally I'd like to do this without transcoding the incoming data. I also want to be able to later re-encode the media with AVAssetExportSession if the user chooses to do so.
Because I want to create a file that is compatible with iOS, I'm limited (I believe) to mpeg, mp4 or quicktime (mov) formats.
Whenever I try to use one of these formats, I see the following warnings during my call to avformat_write_header:
[mov # 0x16401c00] Codec for stream 0 does not use global headers but container format requires global headers
[mov # 0x16401c00] Codec for stream 1 does not use global headers but container format requires global headers
My understanding is that the header wants to know the ultimate file size, which I do not know (the RTSP server is live streaming a camera, and the user stops the recording whenever they want). I guess that makes sense, but I know that others have successfully done this using the ffmpeg command line, so I'm confused as to what else I need to do here.
If I ignore the warning, I can still proceed with writing the file. If I choose mpeg or mp4 formats, my app crashes when I call av_write_trailer. If I use mov, I can successfully close the file, and the file does play back, but usually fails when I try to hand it to the AVAssetExportSession.
I would appreciate any insight into this. Thanks.
Frank
I found what appears to be a solution -- at least, it eliminates the warning. I had to set the CODEC_FLAG_GLOBAL_HEADER on both the audio and video codecs, before calling avcodec_open2.
I'm developing some Video Editing Apps on Android.
the objective of the app is "Editing Videos on Android".
and...
I'm just completed making video file using some images.
but.. I can't attach audio into the video.
my method is same as follows.
1.VideoStream, audio stream creation using AVFormatContext
2.Movie encoding in video stream was successful
3.Encode codec open in audio stream was successful
4.Set sample format to AV_SAMPLE_FMT_FLTP
5.Sample rate and channel was set same as source audio
6.Choose appropriate Decoder and read packet
7.Convert packets using swr_converter, setting same as sample format
8.Encode converted data
9.memory deallocation
10.END!
Problem is here:
Video of finally created video file was normally played. but the Audio wasn't.
It heared like weird. It have many noises and plays slowly.
I've googled with many keywords but they only say about "FFmpeg command line usage".
I wanna make with FFMpeg API. not a Command line tool.
Please help.
Your question is vague without some kind of code to go along with it, as trust me there are a lot of things that can go wrong when using ffmpeg's libraries directly (and on Windows there is no debuging). Unfortunately ffmpeg's libraries are not well documented so it is generally best to read the source code for ffmpeg in order to use its libraries. Find the equivalent command line options to perform what you want and track that through ffmpeg's source to see the library calls.
I want to play a smooth streaming (.ims) file with smooth streaming sample player on my local machine on which i don't have server.
I read that you should give a url of the file in the InitParams param section in the html of the player. I tried with URL like file://localhost/D:/SmoothStreamin/A_MSS_1280_720p_24fps_200kbps/A_MSS_1280_720p_24fps_200kbps.ism but it didnt worked.
file://localhost/D:/SmoothStreamin/A_MSS_1280_720p_24fps_200kbps/A_MSS_1280_720p_24fps_200kbps.ism /Manifest
dont forget the manifest part.
i am sending live rtmp stream to wowza server with a live application config but everytime i connect to the stream for watching the live stream, its start from the beginning of the stream. I can see wowza is creating bigger and bigger file in /content directory and this file will everytime be played from the beginning.
How can i say wowza to send it live, like send the last 10 seconds of the file?
Best regards,
Chris
You'll probably have more luck asking this question on the Wowza forums, where their support team regularly addresses these questions. You'll need to provide more information:
What is your input (camera, flash media encoder, file)?
Can you stream VOD?
A large file building in the content directory sounds like you may be recording the stream. Do you have any add-ons such as the live stream recorder installed?
What url are you using to connect?
Well that answer is simple. You set the wrong stream type in the config.
I guess you set rtp-live-record instead of rtp-live that should fix it.
Kind Regards, Sui
Follow these instructions to create a live stream and broadcasting it
Go to the Wowza Media Server Directory. Probably it is /usr/local/WowzaMediaServer
cd applications;mkdir live
Here live is your application name
cd ../conf;mkdir live;cp Application.xml live/
Now edit the Application.xml file
cd live;vim Application.xml
Change the stream type default to live
live
set HTTPStreamers
cupertinostreaming,smoothstreaming,sanjosestreaming
set LiveStreamPacketizers to
cupertinostreamingpacketizer, smoothstreamingpacketizer, sanjosestreamingpacketizer
set Playmethod
none
Now go to your browser and open the player
file:///usr/local/WowzaMediaServer/examples/LiveVideoStreaming/client/live.html
Server : rtmp://;1935/live
Stream :
Note * : For live streaming you have to use an encoder ( Adobe Flash media Live Encoder , etc )